Making MVP210/410/810 and Asterisk working together.

There's a few parts envolved:

1. Initial configuration: sadly this has to be done with Multitech's windows only program. Connects the modem cable to MVP810 and an available pc serial port, start the program. It will find the MVP810 automatically.

Once connected, you can enable DHCP, the web interface, snmp etc. This way, any subsequent configuration could be completed via a java enabled browser.

2. Configure the channels. In my case, all 8 channels are FXOs, they are connected to 8 PTSN analog lines.

3. Incoming calls (from asterisk to mvp810 to FXO/PTSN)

Simply forward all calls to the hunt group.

4. Outgoing calls (from PTSN to asterisk)

Need to configure SIP proxy. Then also need to make sure the 'Auto Call' feature is enabled for all lines. This way, calls come in on FXO ports will be forwarded to asterisk automatically.

Tips on how to set MULTITECH OUTGOING CALLS (Incoming from PSTN to Asterisk):

After an afternoon spent on how to make this box working with asterisk I decided to post the following tips:

1. In order to make calls pass through the MultiVoIP and get into asterisk, configure asterisk as a SIP Proxy in "Call Signaling"menu, SIP menu, IP, port, user name and password and activate "use SIP proxy" and "allow incoming calls through SIP Proxy only".

2. In Voice/FAX menu activate "auto call" feature and put at least 2 numbers in "phone number" field eg. "80" - put the codec tomanual and select G711 Alaw codec - disable echo cancellation and silence compression and fax features if you don't need them -do the same for every channel

3. In Interface menu leave the FXO ring count at least at 2 and enable "BellCore" Caller ID feature if you want to make Call IDto be read and pass to asterisk afterward - do the same for every channel

4. In Ethernet/IP parameters change the Gateway Name with 0 for example if you don't wanna read "MultiVoip" or anything else onyour IP phone display as a caller ID everytime you get a call...

7. edit outbound phone book and add an entry with "accept any number" IP "0.0.0.0", something for description, enable "SIP proxy"UDP 5060 and IP of your asterisk gateway as a SIP URL.

8. save and reboot

9. if you need to put a prefix before the caller ID in order to make redials on your IP phone, just use this string in asteriskin the incoming context (this scenario is for 0 as a prefix number to dial through PSTN and incoming calls going to extension100):

Incoming calls will get through asterisk and the telephone will ring after 2 rings inside the multivoip only. This is because of

the FXO rings count value set to 2. If this parameter is set less than 2 the BellCore protocol won't read the Caller ID.

If you need to use the multivoip as a sip trunk in FreePBX, create one which registers as <any number>@<your multivoip> with no password and some random number as an outbound dial prefix.

Then configure the multivoip in "SIP Server->Configuration" to accept undefined registrations and accept them from asterisk's IPThen create an entry on the multivoip in the "Phone book -> inbound Phone Book" with the remove prefix the same as your outbound dial prefix.

After that you just need an oulbound route in FreePBX.

Making MVP210/410/810 and Asterisk working together.

There's a few parts envolved:

1. Initial configuration: sadly this has to be done with Multitech's windows only program. Connects the modem cable to MVP810 and an available pc serial port, start the program. It will find the MVP810 automatically.

Once connected, you can enable DHCP, the web interface, snmp etc. This way, any subsequent configuration could be completed via a java enabled browser.

2. Configure the channels. In my case, all 8 channels are FXOs, they are connected to 8 PTSN analog lines.

3. Incoming calls (from asterisk to mvp810 to FXO/PTSN)

Simply forward all calls to the hunt group.

4. Outgoing calls (from PTSN to asterisk)

Need to configure SIP proxy. Then also need to make sure the 'Auto Call' feature is enabled for all lines. This way, calls come in on FXO ports will be forwarded to asterisk automatically.

Tips on how to set MULTITECH OUTGOING CALLS (Incoming from PSTN to Asterisk):

After an afternoon spent on how to make this box working with asterisk I decided to post the following tips:

1. In order to make calls pass through the MultiVoIP and get into asterisk, configure asterisk as a SIP Proxy in "Call Signaling"menu, SIP menu, IP, port, user name and password and activate "use SIP proxy" and "allow incoming calls through SIP Proxy only".

2. In Voice/FAX menu activate "auto call" feature and put at least 2 numbers in "phone number" field eg. "80" - put the codec tomanual and select G711 Alaw codec - disable echo cancellation and silence compression and fax features if you don't need them -do the same for every channel

3. In Interface menu leave the FXO ring count at least at 2 and enable "BellCore" Caller ID feature if you want to make Call IDto be read and pass to asterisk afterward - do the same for every channel

4. In Ethernet/IP parameters change the Gateway Name with 0 for example if you don't wanna read "MultiVoip" or anything else onyour IP phone display as a caller ID everytime you get a call...

7. edit outbound phone book and add an entry with "accept any number" IP "0.0.0.0", something for description, enable "SIP proxy"UDP 5060 and IP of your asterisk gateway as a SIP URL.

8. save and reboot

9. if you need to put a prefix before the caller ID in order to make redials on your IP phone, just use this string in asteriskin the incoming context (this scenario is for 0 as a prefix number to dial through PSTN and incoming calls going to extension100):

Incoming calls will get through asterisk and the telephone will ring after 2 rings inside the multivoip only. This is because of

the FXO rings count value set to 2. If this parameter is set less than 2 the BellCore protocol won't read the Caller ID.

If you need to use the multivoip as a sip trunk in FreePBX, create one which registers as <any number>@<your multivoip> with no password and some random number as an outbound dial prefix.

Then configure the multivoip in "SIP Server->Configuration" to accept undefined registrations and accept them from asterisk's IPThen create an entry on the multivoip in the "Phone book -> inbound Phone Book" with the remove prefix the same as your outbound dial prefix.